similar to: error installing Twinkle - libresolv.so.2(GLIBC_PRIVATE)

Hi, I hope it's the right list. Anyway, I'm using CentOS 5.2. I wrote a program that uses rtkaio (so my Makefile contains -lrtkaio). The program runs successfully. The problem began when I wanted to create a rpm for my program. I was able to generate a binary rpm, but when I tried to install it, I got the following error: [root at klab002 tmp]# rpm -i /tmp/mypkg-1.0.x86_64.rpm error:

I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no

Hi, I have a project that is a library that links against libresolv, It works fine on recent distros: Ubuntu 10.x Fedora 13, Mandriva 2010.1 but on Centos 5.x I get the following errors glibc installed is: glibc-2.5-18.el5_1.1 g++ -DHAVE_CONFIG_H -I. -I./include -I/usr/include/postgresql -O3 -ansi -Wall -Wno-deprecated -D_FORTIFY_SOURCE=0 -MT testUpLog.o -MD -MP -MF .deps/testUpLog.Tpo

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Author: anay Repository: /hg/zfs-crypto/gate Revision: ecd424d223eba74f36389e19f048782a9ce9142e Log message: 6274683 gcc and libresolv[2] don''t get along Files: update: usr/src/lib/libresolv/res_comp.c update: usr/src/lib/libresolv/res_debug.c update: usr/src/lib/libresolv/res_gethost.c update: usr/src/lib/libresolv/res_init.c update: usr/src/lib/libresolv/res_mkquery.c update:

Anyone have an rpm for twinkle? http://www.twinklephone.com/ A new sip softphone.

The Gnu Telephony site: http://wiki.gnutelephony.org Has a Centos repo: http://dist.gnutelephony.org/RPMS/ But I caught some text stating that this is for Centos 4.2. Is it really? Is there a difference; i.e. would it be safe to install these on Centos 4.4? Really I am after Twinkle, and it seems there is a lot you need to actually get Twinkle installed...

https://bugzilla.samba.org/show_bug.cgi?id=2802 Summary: rsync uses extraneous libresolv.dylib Product: rsync Version: 2.6.5 Platform: PPC OS/Version: Mac OS X Status: NEW Severity: trivial Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: psfales@lucent.com

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --------------------------------------------------------------------------- 400/400

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Hi what do you guys use for shared calendars these days? I've been using corporate time but that is now part of Oracle suite so looking for an alternative that is hopefully open source/free to complete my 'get away from exchange' package. thanks

Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Hi, I have already installed 4.2 on my laptop, I only chose GNOME at install time. I now need to use up2date to install kde (I wish to install twinkle softphone). What is the correct way to do that. I did: up2date kde* up2date -u kde* up2date -i kde* It printed a listing of the packages but did not install them. Thanks, Jerry

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status